Recent advances in signal processing technology have allowed the development of new products. One product is the full-duplex speakerphone. Prior technology only allowed half-duplex operation because of the proximity between the loudspeaker and the microphone caused positive feedback and echo. However, half-duplex speakerphones are annoying to users because the speakerphone output is muted while the speaker is talking. The party at the other end is unable to interrupt the conversation until the speaker is quiet for a given length of time.
However, signal processing technology is able to measure room acoustics and automatically cancel any echo thereby generated. The signal processor typically uses an adaptive finite impulse response (AFIR) filter whose coefficients are weighted in accordance with the room acoustics. Each AFIR filter coefficient is multiplied by an audio input signal sample which is delayed by a predetermined number of samples from the current input signal sample. For example if the room causes an echo 50 milliseconds (ms.) after an input signal, the AFIR filter coefficients for samples delayed by 50 ms. are set to cancel this echo. Thus, the signal processor is able to cancel out the echo.
However, full-duplex speakerphones using present signal processing technology have noise problems. One problem is that the echo cancellation process produces noise during operation. Since echoes may be generated up to several hundred ms. after the input signal for some environments, full-duplex speakerphones typically must implement very large AFIR filters. For example, full-duplex speakerphones typically require approximately 1000 AFIR filter taps for small rooms. More-complex speakerphone systems, such as teleconferencing systems for larger rooms having multiple microphones and speakers, may require as many as 4000 taps. Since the speakerphones must be able to operate in a variety of environments, they are designed to accommodate environments having high levels of echo. However, noise increases as the numbers of taps increases, resulting in unnecessary noise increases for small room environments.
Another problem is initialization noise, which is more acute for complex systems such as teleconferencing systems. Before normal operation, the system must initialize the AFIR filter coefficients according to the room acoustics to determine all echo paths. After initialization, the coefficients may be continuously updated. A typical teleconferencing system requires five to twenty seconds of initialization upon power up. During this initialization sequence, the loudspeakers broadcast noise, typically white noise, in order to measure the echo characteristics. Another technique generates a chirp signal instead of white noise. The signal processor generates the chirp by rapidly sweeping all the way from a very low frequency to the Nyquist frequency. Either type of initialization is very annoying to users. Furthermore, the user cannot keep the system continually operational because the adaptive echo cancellation filter coefficients diverge from optimum during long periods of silence.